Log - c21988f4233db75deceaaa9800207bcc7d35174e - platform/external/webrtc

c21988f Remove codereview.settings by henrik.lundin@webrtc.org · 13 years ago
e12b1b5 Revert 3428 by bjornv@webrtc.org · 13 years ago
61ec7da Delay estimator wrapper API changes. This should finalize the changes to delay estimator making it work for multi-probe. by bjornv@webrtc.org · 13 years ago
57e6b81 Mac 64-bit compatibility for WebRTC. by henrike@webrtc.org · 13 years ago
d94659d Initial upload of NetEq4 by henrik.lundin@webrtc.org · 13 years ago
63e0964 Fix webrtc compilation errors for Chrome Win64 by andrew@webrtc.org · 13 years ago
9ae4c66 Set working dir for test run script + update resources by kjellander@webrtc.org · 13 years ago
e1888af Add <(DEPTH) to global includes by kjellander@webrtc.org · 13 years ago
bf535b9 Optimize NACK list creation. by stefan@webrtc.org · 13 years ago
b2d7497 Fix Win64 warnings by kjellander@webrtc.org · 13 years ago
8526459 Added tests for multiple near-end support. by bjornv@webrtc.org · 13 years ago
57f3a11 Short CL: only name change. by bjornv@webrtc.org · 13 years ago
94c213a Separated far-end handling in BinaryDelayEstimator. by bjornv@webrtc.org · 13 years ago
59d2095 Moving ViE test files and deleting files no longer used. by mflodman@webrtc.org · 13 years ago
d3ecb61 Fix path to perf Python scripts in test.gyp by kjellander@webrtc.org · 13 years ago
43da54a Reformatted rtp_sender: made lint clean. by phoglund@webrtc.org · 13 years ago
3e47a0a Test launching script by kjellander@webrtc.org · 13 years ago
c4373bc Moved several function pointer declarations in iSAC to isac initialization file. by kma@webrtc.org · 13 years ago
16d540e Fixed text relocation code related to ARM assembly code. by kma@webrtc.org · 13 years ago
e8482f0 Revert 3406 by kma@webrtc.org · 13 years ago
cd2f135 Revert 3405 by niklas.enbom@webrtc.org · 13 years ago
ebef7e4 Moved all function pointer declarations in iSAC to a single place. by kma@webrtc.org · 13 years ago
05e7bfe Mainly hlundin's patch. by niklas.enbom@webrtc.org · 13 years ago
4782911 Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. by kma@webrtc.org · 13 years ago
5dfb1f2 Bug fix in WebRtcOpus_DurationEst by henrik.lundin@webrtc.org · 13 years ago
8126602 Fix frame_editing_unittest.cc by kjellander@webrtc.org · 13 years ago
a812a3a Updated version number to 3.21 by elham@webrtc.org · 13 years ago
0973861 Fixes payload spelling error. by henrike@webrtc.org · 13 years ago
5accd37 RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. by phoglund@webrtc.org · 13 years ago
8382ad5 Added perf expectations for stack tests. by phoglund@webrtc.org · 13 years ago
ae1a58b Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 13 years ago
899699e Enabled full lint checking for ALL WebRTC changes. by phoglund@webrtc.org · 13 years ago
a678a3b Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. by stefan@webrtc.org · 13 years ago
a3c82bf Remove '<(library)' in gyp files. by wjia@webrtc.org · 13 years ago
bb599b7 This CL includes part of changes in a larger one. The final goal is to allow multiple delay estimators using the same reference (far-end) to save computational complexity. by bjornv@webrtc.org · 13 years ago
a2d8b75 An API to get the internal estimation quality in the delay estimator has been added. Unit tests have been updated. There is no impact to other parts in WebRTC. by bjornv@webrtc.org · 13 years ago
2e2a4cf Remove <(library) from gyp file. by wjia@webrtc.org · 13 years ago
a3e6bec Posix Thread: Removes the setting of the run function to NULL which could cause data race. by henrike@webrtc.org · 13 years ago
4ad6445 Fixed URL unquoting in bot names. Added iOS Device. Removed unnecessary filter code. by phoglund@webrtc.org · 13 years ago
c39962a Adding TRYSERVER_ROOT to codereview.settings by kjellander@webrtc.org · 13 years ago
218c542 Make VoE handle longer delays by niklas.enbom@webrtc.org · 13 years ago
c7e935f Adding timeEndPeriod to Synchronize function, see bug for details. by mflodman@webrtc.org · 13 years ago
efae5d5 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead. by phoglund@webrtc.org · 13 years ago
20ed36d Break out RtpClock to system_wrappers and make it more generic. by stefan@webrtc.org · 13 years ago
3b7feb2 Convert psnr and ssim to strings before printing them. by stefan@webrtc.org · 13 years ago
a4b5886 Add a counter to the video rtp play output filename. by stefan@webrtc.org · 13 years ago
ebc6d8f libyuv r540 roll for valgrind tools update, optimized ARGBToI444_SSSE3 and I420Copy single memcpy per plane if contiguous. by fbarchard@google.com · 13 years ago
00c18db Fix libvpx for Android by hclam@chromium.org · 13 years ago
2fd947f Removing outdated comment by mikhal@webrtc.org · 13 years ago
14d1898 Removing arena_thread_freeres suppression by kjellander@webrtc.org · 13 years ago
acfdd96 Reformatted rtp_rtcp_impl*. by phoglund@webrtc.org · 13 years ago
77a584b Made ViEToFileRenderer use a separate thread for rendering frames to file. by stefan@webrtc.org · 13 years ago
a22a9bd Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. by phoglund@webrtc.org · 13 years ago
49273ff logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid() by braveyao@webrtc.org · 13 years ago
b119369 Fix android clang build. by wjia@webrtc.org · 13 years ago
3f9db37 Fix android clang build. by wjia@webrtc.org · 13 years ago
bafdae3 Fix simulated analog gain in audioproc. by andrew@webrtc.org · 13 years ago
f908011 Remove extra line. by andrew@webrtc.org · 13 years ago
75ba519 Updating chromium_revision 169394:176094 by kjellander@webrtc.org · 13 years ago
e7dc7f8 Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM. by stefan@webrtc.org · 13 years ago
26901c2 libyuv r534 for tools folder valgrind and endian fix for big endian platforms like s390x. by fbarchard@google.com · 13 years ago
be86a6d Explicitly disable sincos optimization on Android. by leozwang@webrtc.org · 13 years ago
e468f08 Disable PSNR/SSIM thresholds for the Gilber-Elliot test. by stefan@webrtc.org · 13 years ago
171ac59 Corrected TSAN suppression. by phoglund@webrtc.org · 13 years ago
dc6fa02 Fixing error in argument parsing by kjellander@webrtc.org · 13 years ago
8f13810 Improved memory tool test wrapper script by kjellander@webrtc.org · 13 years ago
0af0d3d Address a build issue with Android-Clang compiler: by kma@webrtc.org · 13 years ago
ef1a760 Rounding error fix in media_opt_util. by marpan@webrtc.org · 13 years ago
a5e7e76 Use %d for signed value in trace. by andrew@webrtc.org · 13 years ago
08d660f Allow for some error in volume testing. by andrew@webrtc.org · 13 years ago
d005468 Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. by phoglund@webrtc.org · 13 years ago
2f225ca Add logs when no RTCP RR has been received for three regular RTCP intervals. by mflodman@webrtc.org · 13 years ago
d66eb8c Disabled GQoS since it breaks ViE auto test. by henrika@webrtc.org · 13 years ago
fcd8585 Enable external encoders with internal picture source. by stefan@webrtc.org · 13 years ago
658d423 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers. by mikhal@webrtc.org · 13 years ago
27cb301 Updated version number to 3.20 by elham@webrtc.org · 13 years ago
bc9a959 Generalized suppression for Trace::Add by phoglund@webrtc.org · 13 years ago
acc54b4 Added perf expectations and corrected existing tests to remove spaces from series names. by phoglund@webrtc.org · 13 years ago
c38eef8 Reformatted RTPReceiver. by phoglund@webrtc.org · 13 years ago
df3a15f Removed spaces from full stack test labels, consolidated graphs by phoglund@webrtc.org · 13 years ago
1ea4b50 Refactor receiver.h/.cc. by stefan@webrtc.org · 13 years ago
1926d33 Change Sleep() comment in test fixture. by andrew@webrtc.org · 13 years ago
bcb7174 .gitignore: Add *.mk, created as part of ChromiumOS build by andrew@webrtc.org · 13 years ago
f545cf8 Addressing webrtc issue 1237, http://code.google.com/p/webrtc/issues/detail?id=1237. by kma@webrtc.org · 13 years ago
91d8933 Dashboard LKGR parsing builder names by kjellander@webrtc.org · 13 years ago
6f62836 Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?) by phoglund@webrtc.org · 13 years ago
5c8d9d3 Reformatted tick_util. by phoglund@webrtc.org · 13 years ago
daabfd2 Reformatted trace* files. by phoglund@webrtc.org · 13 years ago
201d4b6 Fix implicit conversion error in mixing test. by andrew@webrtc.org · 13 years ago
b2b628d Further relax thresholds in mixing test. by andrew@webrtc.org · 13 years ago
00c7c43 Replace voice engine utility functions with system wrapper variants. by andrew@webrtc.org · 13 years ago
943770b Fixed various problems with the reformat script: by phoglund@webrtc.org · 13 years ago
ec9c942 Reformatted thread and static_instance. by phoglund@webrtc.org · 13 years ago
a19d04e Coverity now uses Visual Studio 2010 project file by kjellander@webrtc.org · 13 years ago
1b6da28 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. by pwestin@webrtc.org · 13 years ago
f556890 Added possibility to repeat frames. Also added unittest for that feature. by brykt@google.com · 13 years ago
d73527c Changed assert to log. by mflodman@webrtc.org · 13 years ago
d0d4149 Adding AUDIO application as default for Opus stereo by tina.legrand@webrtc.org · 13 years ago
ad0ed58 Fixed a missed initialization (found by valgrind FYI bot). by phoglund@webrtc.org · 13 years ago
ac77084 Roll opus to 172355 and delete opus_demo from webrtc opus by leozwang@webrtc.org · 13 years ago